Voice dialing methods and apparatus implemented using AIN techniques

ABSTRACT

Methods and apparatus for implementing communication services such as voice dialing services are described. In one Centrex based voice dialing embodiment, voice dialing service subscribers are given access to personal voice dialing records including calling entries via the Internet as well as via telephone connections. Each calling entry normally includes the name and, optionally nickname, of a party to be called. It also includes one or more telephone numbers associated with each name. Different telephone number identifies, e.g. locations, can be associated with different names. A user can create or update entries in a voice dialing directory using text conveyed over the Internet or speech supplied via a telephone connection. In order to facilitate updating and maintenance of voice dialing directories over the Internet speaker independent (SI) speech recognition models are used. When a calling entry is created via the Internet the text of the name is processed to generate a corresponding speech recognition model there from. When an entry is created via speech obtained over the telephone, a speech recognition model is generated from the speech and a text name is generated is generated using speech to text technology. To avoid having to hang-up and initiate a new voice dialing call the outcome of a voice dialing call is monitored and the subscriber is provided the opportunity to initiate another call using voice dialing if the first call did not complete successfully e.g., goes unanswered.

RELATED APPLICATIONS

This application claims the benefit of U.S. Provisional ApplicationSerial No. 60/180,640 filed Feb. 7, 2000.

FIELD OF THE INVENTION

The present invention is directed to communications systems and, moreparticularly, to methods and apparatus for providing voice dialingservices.

BACKGROUND OF THE INVENTION

Telephones, both mobile and land based, are a frequently usedcommunications tool of modern society. While basic telephone service hasremained generally unchanged in terms of its features for years, thereis an ever increasing demand for new telephone services.

The demand for new telephone services is prompted by a desire to rendertelephones easier to use and/or desires to make them more efficientcommunication tools. The demand for new telephone services is alsofueled by the desire of individual telephone companies to; distinguishthe services they offer from those of their competitors; create newrevenue sources; and/or expand existing revenue sources.

In order to provide enhanced telephone services, many telephonecompanies now implement a telephone communications network as anAdvanced Intelligent Network (AIN) which has made it easier to provide awide array of previously unavailable voice grade telephone servicefeatures. In an AIN system, telephone central offices, each of whichserves as a signal switching point (SSP), detect one of a number of callprocessing events identified as AIN “triggers”. An SSP which detects atrigger suspends processing of the call which activated the trigger,compiles a call data message and forwards that message via a commonchannel interoffice signaling (CCIS) link to a database system, such asa Service Control Point (SCP). The SCP may be implemented as part of anintegrated service control point (ISCP). If needed, the SCP can instructthe central office (SSP) at which the AIN trigger was activated toobtain and forward additional information, e.g., information relating tothe call. Once sufficient information about the call has reached theISCP, the ISCP accesses stored call processing information or records(CPRs) to generate from the received message data, a call controlmessage. The call control message is then used to instruct the centraloffice on how to process the call which activated the AIN trigger. Aspart of the call control message, an ISCP can instruct the centraloffice to send the call to an outside resource, such as an intelligentperipheral (IP) using a send to outside resource (STOR) instruction. IPsare frequently coupled to SSPs to provide message announcementcapabilities, voice recognition capabilities and other functionalitywhich is not normally provided by the central office. The controlmessage is normally communicated from the ISCP to the SSP handling thecall via the CCIS link. Once received, the SCP completes the call inaccordance with the instructions received in the control message.

One service which can be implemented with AIN functionality is Wide AreaCentrex. Centrex takes a group of normal telephone lines and providescall processing to add business features to the otherwise standardtelephone lines. For example, Centrex adds intercom capabilities to thelines of a specified business group so that a business customer can dialother stations within the same group, e.g., lines belong to the samecompany, using extension numbers such as a two, three, or four digitnumbers, instead of the full telephone number associated with eachcalled line. Other examples of Centrex service features include calltransfer between users at different stations of a business group and anumber of varieties of call forwarding. Thus, Centrex adds a bundle ofbusiness features on top of standard telephone line features withoutrequiring special equipment, e.g., a private branch exchange (PBX) atthe customer's premises. U.S. Pat. No. 5,247,571, which is herebyexpressly incorporated by reference, describes in detail a Wide AreaCentrex system implemented using AIN techniques.

Voice dialing is a useful service which has been implemented in someknown systems by having control logic in a telephone switch connect acaller to a voice dialing IP which provides the voice dialing service.The telephone switch may couple the subscriber to the voice dialing IPas a result of the subscriber calling a telephone number correspondingto the IP or entering a code which is detected by the switch. Such knownvoice dialing systems are not AIN based and therefore are somewhatlimited in terms of the logic and information available for controllingconnections to voice dialing apparatus, e.g., voice dialing IPs. Thus,the known techniques of using logic embedded in a switch to determinewhen and to which voice dialing IP a caller should be connected can leadto inefficient using of voice dialing IP resources and limit the abilityof known voice dialing services to be implemented as an integral part ofother services, e.g., Centrex Services.

Voice dialing is a particularly desirable service since it eliminatesthe requirement that a user of the voice dialing service remember thetelephone number of the party being called. In various known voicedialing systems, speaker dependent speech recognition is used toidentify spoken names. In such systems, a personal dialing directory ismaintained for each subscriber of the voice dialing system. The personaldialing directory is a database which includes a speaker dependentspeech recognition template for each of a plurality of names which maybe spoken and a telephone number for each name. When used, the data,e.g., templates in a subscriber's directory, are retrieved; a speakerdependent speech recognition operation is performed on a spoken nameprovided by the subscriber; and then, assuming a name is recognized, thecall is completed to the telephone number in the subscriber's personaldialing directory corresponding to the name. An. IP may be used forperforming the speech recognition and various other tasks associatedwith the known voice dialing operation. One particular system forimplementing voice dialing is described in detail in U.S. Pat. No.5,832,063.

While voice dialing systems which use speaker dependent speechrecognition to identify spoken names enjoy a high degree of recognitionaccuracy, they have the disadvantage of requiring that a user of thesystem provide one or more utterances of each name for which speakerdependent speech recognition templates are to be generated. Thus, avoice signal, e.g., voice telephone connection, is normally requiredwhen adding or updating names in a personal dialing directory. Inaddition, the need to provide multiple utterances of each name in thepersonal dialing directory can prove irritating to some customers.

In an attempt to make services provided using AIN techniques easier tomanage, management of AIN services such as, e.g., call forwarding, via apersonal computer and the Internet have been suggested. U.S. Pat. No.5,958,016, which is hereby expressly incorporated by reference,describes a system wherein a web page type interface is provided, whichallows a subscriber access to control and reporting functionalities ofan AIN system via the Internet.

Unfortunately, the use of speaker dependent speech recognitiontemplates, with the corresponding need for multiple speech samples totrain each name in a personal dialing directory, has made Internet basedmanagement of existing voice dialing systems of the type described abovedifficult to implement.

While existing Centrex and voice dialing services are useful, it isdesirable that such services continue to be improved and enhanced. Withregard to voice dialing, it is desirable that new methods and apparatusbe devised which would allow for Internet based management of voicedialing services. It is also desirable that new methods of providingvoice dialing services be devised which will allow voice dialingservices to be implemented as part of AIN based service packages such aCentrex. With regard to Centrex, it is desirable that Centrex service beenhanced to support voice dialing functionality as well as Internetbased management of said functionality.

SUMMARY OF THE INVENTION

The present invention is directed to methods and apparatus which can beused to provide voice dialing, enhanced Centrex, and othercommunications services. In various embodiments, the communicationsservices of the present invention are implemented to facilitate easymanagement of the services by end users via the Internet. In accordancewith one feature of the present invention, voice dialing is implementedas an AIN based service. By implementing voice dialing as an AIN basedservice, the service can be easily integrated with Centrex and/or otherAIN based services. In addition, since control logic in an ISCP is usedto control access to voice dialing hardware, in response to activationof one or more AIN triggers set telephone switches, access to voicedialing hardware can be controlled to provide efficient use of the voicedialing hardware regardless of the location from which a voice dialingservice subscriber calls. In addition, ISCP logic can translate multipleidentifiers, e.g., various telephone numbers, associated with asubscriber, into a single user identifier which can be used by voicedialing hardware for subscriber information retrieval purposes.

In accordance with an exemplary voice dialing embodiment of the presentinvention, each subscriber is able to maintain and update a voicedialing record, used to provide voice dialing services via the Internet.The subscriber's voice dialing record, sometimes refereed to as a voicedialing directory, may include such information as a user identifier,e.g., a Centrex telephone number associated with the subscriber, amobile telephone number associated with the subscriber, an additionaltelephone number used by the subscriber, information on one or moreindividuals or parties to be called and, optionally, informationidentifying a corporate voice dialing record to be used if a spoken nameor nickname is not found using speech recognition models included in thesubscriber's voice dialing record.

In accordance with the present invention, a voice dialing call may beplaced to an individual or party by speaking either the name or anickname of the individual or party being called. In addition, ifdesired, a location associated with the party or individual may also bespecified in addition to the party or individuals name. When a locationis stated in conjunction with a name, the call will be placed to theparty or individual at the particular specified location.

To support such voice dialing functionality, a voice dialing record ismaintained for each subscriber. Information relating to each party orindividual who may be called is stored within the subscriber's voicedialing record in the form of a calling entry. The subscriber isresponsible for providing and updating the information included in eachcalling entry. A calling entry normally includes the name of the partyor individual who may be called, an optional nickname for the party, anda first telephone number associated with the named party or individual.Optionally one or more additional phone numbers may be associated in thecalling entry with the named party or individual. When multipletelephone numbers are associated with a named party or individual, atelephone number identifier, e.g., the name of a location, is normallyassociated in the calling entry with each telephone number. Examples oftelephone number identifiers which may be associated with a telephonenumber include, for example: Home, Office, Office 2, Mobile, Secretary,etc. In some embodiments the subscriber is allowed to define telephonenumber identifiers. For example, the subscriber may associate theidentifier “girl friend 2” with the telephone number of a second girlfriend.

A subscriber's voice dialing record is stored in part of the publictelephone system in accordance with the present invention, e.g., in anintelligent peripheral device, e.g., voice dialing (VD) IP, coupled to acentral office telephone switch. In addition to being coupled to thetelephone network via the central office switch, the VD IP is coupled tothe Internet via one or more servers. Various security measures aretaken, including the use of a PIN, to insure that unauthorizedindividuals, e.g., non-subscriber's, are not given access via theInternet, to subscriber's voice dialing records. Alternatively, acustomized desk top application, as opposed to a Web Browser, can beused to access and update a subscriber's voice dialing recordinformation via the Internet. Given that the VD IP is coupled to boththe telephone network and the Internet, two methods of accessing asubscriber's records are possible.

To provide a subscriber with the greatest flexibility with regard tomaintaining and updating his/her voice dialing directory, a subscriberis provided the opportunity to update the subscriber's voice dialingdirectory by telephone using voice/DTMF input and, alternatively, thoughan Internet connection. A Web browser such as Internet Explorer,operating on a subscriber's computer can be used to display voicedialing record information and for providing updated information, e.g.,in the form of text input, to the VD IP in which the subscriber's voicedialing record is stored.

In addition to the calling record, e.g., calling entry informationdiscussed above, which is provided by the subscriber, each subscriber'scalling record normally further comprises a speech recognition model foreach name and nickname included in the subscriber's record. The use ofspeaker dependent speech recognition models for the recognition of oneor more names included in a voice dialing record is contemplated andpossible. However, in order to implement a voice dialing system that iseasy to update and maintain via text obtained from the Internet, speakerindependent speech recognition models are used for most, if not all,names and nicknames included in a subscriber's voice dialing record. Thespeech recognition models for names entered via the Internet aregenerated from the text of the name using speaker independent modelingtechniques. This generally involves performing a text to phonemeconversion operation. It also involves generating a speaker independentspeech recognition model from the phonemes produced from the text of thename or nickname being processed.

Speaker independent speech recognition models produced from text workwell for most names. However, in some cases where the pronunciation of aname is difficult to predict from its spelling, e.g., due to a varietyof possible pronunciations or because it originates from a foreignlanguage name, a speaker independent speech recognition model producedfrom text may provide recognition results which are less thansatisfactory.

In order to address this potential problem, the methods and apparatus ofthe present invention allow for a subscriber to provide one or morespeech samples of a name to be used for generating a speech recognitionmodel for voice dialing purposes. The speech corresponding to a name maybe provided as part of a telephone based voice dialing entry creation orupdating process. In the case where an existing entry is being updatedusing a spoken version of a name, a speaker independent (or, optionallyspeaker dependent) speech recognition model is generated from the speechcorresponding to the name. The generated speech recognition model isstored in place of a speech recognition model previously generated fromtext when such a text based model exists.

In the case where speech corresponding to a name is being used to createa voice dialing entry, as opposed to update an existing entry, thespeech corresponding to the name is used to generate a speechrecognition model. A speech recognition operation is also performed onthe speech and a text version of the spoken name is generated and usedto populate the (text) name portion of the calling entry correspondingto the spoken name. The text name entry is displayed to a user accessingthe calling entry information via the Internet. A user can edit, via theInternet, the spelling of the name in the entry if desired withoutaffecting the speech recognition model generated from the suppliedspeech.

To distinguish between speech recognition models generated from speechand those generated from text, model type information, e.g., identifyingwhether the model was generated from text or speech, may be stored inconjunction with the speech recognition models included in a voicedialing directory. In one particular embodiment, altering of the textversion of a name normally does not affect the corresponding speechmodel when the model was generated from speech. However, in such anembodiment altering of the text version of a name does cause updating ofthe speech model when the model is generated from text.

As a result of the above discussed model generation processes, asubscriber's voice dialing record may include name speech recognitionmodels generated from text as well as ones generated from speechsamples.

The information in a subscriber's voice dialing record is used by thevoice dialing IP to initiate calls in response to a subscriber's speech.As part of a voice dialing operation information from a subscriber'svoice dialing record is retrieved from memory using a user, e.g.,subscriber, ID to identify the subscriber record. In one embodiment, thesubscriber s Centrex telephone number is used as the User ID. Thus, incases where the subscriber attempts to place a call using speech fromhis/her Centrex phone line, the subscriber's ID can be obtained usingautomatic number identification (ANI) techniques.

In order to allow a subscriber to call from any one of a plurality oftelephones various procedures for identifying the caller and providingthe voice dialing IP with the subscriber's ID are supported.Accordingly, a voice dialing service subscriber using a mobile phone,phone associated in the customer record with the subscriber, or anotherphone, can be coupled to the voice dialing IP where his/her voicedialing record is accessible and obtain a voice dialing service therefrom. In most cases where the subscriber initiates a voice dialing callfrom a phone line other than his/her Centrex line, the telephone numberfrom which the person is calling is identified and cross-referenced withstored information associating the identified phone number, e.g., mobilenumber, with a Centrex telephone number. In cases where the subscriberis calling from a number for which there is no stored informationassociating the phone number with a Centrex phone number, the caller isrequested to supply his/her Centrex telephone number. An integratedservice control point (ISCP), accessed in response to activation of anAIN trigger, may be used for associating telephone numbers from which asubscriber is calling with his/her Centrex number which is used as theuser ID supplied to the voice dialing IP.

The voice dialing IP receives from the subscriber speech which is to beused to place a voice dialing call. The speech may include the name ornickname of a person to be called. It can also include a locationassociated with the name or nickname of the person or party to becalled.

Upon recognizing a name in received speech, the voice dialing IP playsthe voice dialing subscriber a confirmation message such as, e.g.,“Dialing John Smith” where “John Smith” is the recognized name. Thisprovides the subscriber the opportunity to stop the call in the event arecognition error occurred, e.g., by hanging up or otherwise signalingthe voice dialing IP. A voice recording of the recognized name may beplayed to the subscriber as part of the confirmation message, e.g.,recording of the subscriber's voice obtained at the time the subscribertrained a speech recognition model in his/her voice dialing directoryusing speech. However, in accordance with one feature of the presentinvention, a text representation audio corresponding to a recognizedname is generated for message confirmation purposes from a stored textversion of the name. In such an embodiment, a text to speech circuit isused to generate an audio version of a name from a stored text versionwhen an audio version of the name is needed for a confirmation message.Since storage of text is much more efficient than the storage of voicerecordings, the use of text versions of names for confirmation messagepurposes can be considerably more efficient from a memory perspectivethan the use of audio recordings.

In response to recognizing a name or nickname but not a location inspeech, the voice dialing IP provides to the telephone switch to whichit is coupled the default, e.g., first, telephone number, in thesubscriber's voice dialing record, corresponding to the identified nameor nickname. If a location is recognized in the received speech, inaddition to a name or nickname, the telephone number corresponding tothe name or nickname, and the detected location is supplied to thetelephone switch. After supplying the called party telephone numberinformation to the switch and playing the subscriber an audioconfirmation messaged including at least a portion of the recognizedname, the audio connection between the voice dialing IP and the calleris terminated freeing the voice dialing IP resources to be used toservice other subscribers. The telephone number supplied by the VD IP tothe telephone switch is used by the switch to place a call to thespecified number.

In accordance with one embodiment of the present invention, the resultsof a voice dialing operation are monitored to detect various potentialcall outcomes. That is, a determination is made as to whether or not abusy signal is encountered, the called party does not answer, or if thecall is successfully completed to the destination number (e.g. whetherthe called party answers the call).

If a no answer or busy signal condition is detected, a service controlpoint (SCP) associated with the voice dialing subscriber is contactedfor call processing instructions. In various embodiments, the SCP isimplemented as an integrated service control point (ISCP). The contactedSCP then causes the subscriber to be reconnected to the VD IP. The VD IPinforms the caller via an audio message of the call outcome and offersthe caller the opportunity to place another voice dialing call ifdesired.

Thus, the present invention provides a voice dialing subscriber theopportunity to place multiple sequential voice dialing calls withouthaving to hang-up when a busy signal or no answer condition isencountered. Because the voice dialing IP is disconnected from thecaller between voice dialing call attempts, voice dialing and speechrecognition resources are used in an efficient manner and are not tiedup during the time required to detect the outcome of a call.

Various additional features and advantages of the present invention willbe apparent from the detailed description which follows.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 illustrates a communication system implementer in accordance withan exemplary embodiment of the present invention.

FIG. 2 illustrates an exemplary Centrex internet connection accessserver which may be used in the system illustrated in FIG. 1.

FIG. 3 illustrates an exemplary voice dialing intelligent peripheraldevice suitable for use in the system of FIG. 1.

FIG. 4 is a detailed illustration of an exemplary speech recognizerarray which can be used in the voice dialing IP of FIG. 3.

FIG. 5-9 are flow diagrams illustrating steps, messages, and dataassociated with setting up, configuring, and using Centrex accountinformation in accordance with the present invention.

FIGS. 10-14 illustrate various pages which may be displayed to a Centrexvoice dialing service subscriber when the subscriber accesses and/orupdates voice dialing service information via the Internet.

FIG. 15 illustrates an exemplary voice dialing customer record which maybe stored in the voice dialing IP illustrated in FIG. 1.

FIG. 16 illustrates a sub-routine used for updating voice dialingentries based on information supplied over the Internet to the voicedialing IP of FIG. 1.

FIG. 17 illustrates a speech recognition model updating sub-routine usedto generate speech recognition models corresponding to names from actualspeech samples obtained via a telephone connection.

FIGS. 18-20 are call flow diagrams which illustrate the steps, inaddition to the data, instruction and message flow, associated withmaking one or more voice dialing calls in accordance with the presentinvention.

DETAILED DESCRIPTION

As discussed above, the present invention is directed to methods andapparatus for providing voice dialing services. The voice dialingservices may be provided as a stand-alone service, as part of a Centrexservice, or as part of another telephone service package. As will bediscussed in detail below, in accordance with one feature of the presentinvention, individuals can update voice dialing directories maintainedas part of a voice dialing service via the Internet and/or through useof their telephones.

FIG. 1 illustrates a communication system 100 implemented in accordancewith an exemplary embodiment of the present invention. The system 100supports communications via the Internet 30, as well as the publictelephone switching network (PTSN). The PTSN includes a plurality ofsignal switching points (SSPs) 2, 4, 6 which, as is known in the art,may be implemented using known Class V telecommunications switchescapable of supporting the signaling system seven (SS7) protocol. EachSSP 2, 4, 6 may correspond to a different telephone central office.Trunk lines (TL), which may comprise, e.g., one or more T1 lines,interconnect the various SSPs 2, 4, 6. In addition to SSPs 2, 4, 6, thesystem 100 includes a mobile telephone switching office (MTSO) 5 forservicing mobile telephone calls. The calls may be received via antenna7. The MTSO 5 is coupled to one or more SSPs, e.g., SSP 4 via trunklines. In this manner, cellular telephone calls can be routed andprocessed via an SSP 4 allowing interaction with land line telephones asis known in the art.

Each SSP 2, 4, 6 is normally connected to one or more customer premises(CP) which may include, e.g., Centrex subscriber residences and/oroffices as well as the residences and offices of non-Centrexsubscribers. In the FIG. 1 example, first and second customer premises22 and 24 are coupled to the second SSP 4, while third and fourthcustomer premises 26, 28 are coupled to the third SSP 6. Connectionsbetween the SSPs and CPs may be by POTS lines, ISDN lines, DSL, or otherknown communications lines.

Communications equipment, referred to as customer premise equipment(CPE) is located at each customer premises 22, 24, 26, 28. Customerpremise equipment may include, e.g., telephones, faxes, computers, etc.In FIG. 1, a computer 36, land-line telephone 38, and mobile telephone37 are shown as being located at the first customer premises 22. As willbe discussed below, each of these devices corresponds in the exemplaryembodiment shown in FIG. 1 to a first Centrex service subscriber. Sincecell phone 37 is a mobile communications device it need not bephysically located at the first customer premises to operate. Thecomputer 36, located at the first customer premises 22 is coupled by anyone of a plurality of known connection techniques, e.g., telephonedial-up, ISDN, DSL, etc., to the Internet 30, also known as the WorldWide Web.

While the second, third and fourth customer premises 26, 28 areillustrated as including only landline phones, it is to be understoodthat they may have any number of communications devices including, e.g.,telephone, fax, and computer devices. For purposes of explanation, itwill be assumed that the second customer premises 24, represents thebusiness office of the first Centrex service subscriber while the firstcustomer premises 22 represents the residence (home) of the firstCentrex service subscriber. Additional Centrex service subscribers maybe coupled to anyone of the SSPs 2, 4, 6.

The system 100 is implemented using AIN techniques. Accordingly, theprocessing of calls directed to a customer's telephone line and receivedby an SSP from a telephone customer's line may be controlled byinstructions included in customer call processing records (CPAs). In thesystem 100, the CPAs are stored at an Integrated Services Control Point(ISCP) 16. At least one CPR exists for each Centrex service subscriber.A customer's CPR is accessed in response to activation of an AIN triggerset at, e.g., the SSP 2, 4, or 6 to which the subscriber's lines areconnected, e.g., by POTS lines.

The ISCP 16 includes an SCP 64, a service management system (SMS) 62,data and reporting system (DRS) 63, service creation environment (SCE)60, and a network interface (NI) 45. A local network 67 couples thevarious components of the ISCP 16 together.

The network interface 45 couples the ISCP 16 to various other componentsof the telephone network 100 via a TCP/IP based network referred to asan operational services network (OSN) 34. The OSN 34 interconnects SSPs2, 3, 6, the MTSO 5, Intelligent Peripherals (IPs) 18, 10, 20, and theISCP 16. Thus, the OSN 34 is a network over which control and signalinginformation can be passed between the various system components, e.g.,using TCP/IP. In addition to being connected to the OSN 34, ISCP 16 isconnected, via its SCP 64, to the SSPs and MTSO via one or more signaltransfer points (STPs) 12, 14 and Signaling System Seven (SS7)interconnects over which messages, data, and requests for callprocessing control instructions can be communicated between the SSPs 2,4, 6, or MTSO 5 and ISCP 16 in accordance with the SS7 protocol.

The SCP 64 includes a multi-service application platform (MSAP) database69 which includes customer data (CD) 71 for each of a plurality ofCentrex and/or other service subscribers. The customer data 71 includes,for each customer: 1) a list of the services to which the customersubscribes; 2) a password which may be input via DTMF signals; and 3) acall processing record (CPR) which is used to instruct an SSP how toprocess a call in response to an AIN trigger to thereby implement theservices to which the customer subscribes. Exemplary services which maybe supported by the ISCP 16 include, e.g., voice dialing, callforwarding, call screening, voice mail and a host of other serviceswhich may be provided to Centrex subscribers as well as non-Centrextelephone customers.

For purposes of explaining the voice dialing features of the presentinvention, voice dialing services will be described in the context of aCentrex environment. However, it is to be understood that the voicedialing service of the present invention can be provided outside theCentrex environment, i.e., voice dialing services can be provided tonon-Centrex telephone customers as well as Centrex service subscribers.

The customer data 71 which includes call processing records 73 isgenerated, at least initially, by the SCE 60 in response to inputreceived from a service representative or operator 44. Customer data inthe database 71 may, after initial provisioning of a service for acustomer, the CPR may be updated by the customer via the Internet andthe use of a Web browser.

The SCE 60 includes an operator terminal (OT) 49, service orderprocessing circuitry 48 and AIN provisioning system circuitry 46. Theoperator terminal 49 is used by the service representative 44 to enterservice information, e.g., to create a Centrex account for a newsubscriber. The entered data may be information, e.g., relating to theaddition of a new Centrex customer, the adding of a service for anexisting customer, and/or the cancellation of a service being providedto an existing customer. The service order processing circuitry 48 isused to generate service orders, e.g., orders to add or cancel aservice, in response to service information entered into the operatorterminal 49. The AIN provisioning system circuitry 46 is responsible forsetting and/or updating AIN triggers at the various signal switchingpoints (SSPs and MTSO) required to implemented a service order generatedby the service order processing circuitry 48. In addition to setting AINtriggers, the AIN provisioning system circuitry 46 is responsible forgenerating and/or updating customer data, e.g., call processing records73, and other information stored in various locations in the system 100,as required to implement a service order. As will be discussed below,various IPs 10, 18, 20 are used to provide services to Centrex and othertelephone service subscribers. Thus, in addition to updating informationin the customer database 71, the AIN provisioning system circuitry isresponsible for updating information in the various IPs 10, 18, and 20.The updating of the IPs and the setting of AIN triggers can be performedby the AIN provisioning system circuitry 46 through communications withthe various system components conducted using the OSN 34 and/or via 557links.

Once service to a customer has been initially configured by a servicerepresentative 44, a Centrex service subscriber can, in accordance withthe present invention, update various service information though the useof a personal computer and a Web Browser application, various knownbrowsers include Internet Explorer and Netscape. In the FIG. 1 system,the service subscriber to whom the first customer premises 22corresponds can update the subscriber's Centrex information via the useof computer 36 and an Internet connection.

The system 100 includes a Centrex Internet, Customer Access Server(ICAS) 32 which is implemented in accordance with the present invention.The server 32 serves as a secure gateway via which Centrex subscriberscan update and configure their Centrex telephone service informationusing a computer coupled to the Internet. The ICAS 32 includes securityroutines, e.g., a firewall, designed to prevent individuals other thanthe subscriber gaining access to and/or modifying via the Internet,subscriber service information. The ICAS 32 is coupled to the OSN 34thereby allowing a customer, upon satisfying various security checks, toaccess and modify service information stored in any one of the variousnetwork devices, e.g. ISCP 16, and/or IP 18, 10, 20, coupled to the OSN34.

In order to implement various services, such as voice dialing andtelephone access to Centrex customer service information, first, secondand third IPs 10, 18, are used. The first IP 10 is a DTMF IP which iscapable of performing speech recognition and DTMF signal detectionoperations as well as playing voice prompts and other messages to aCentrex customer.

DTMF IP 10 is coupled to the first SSP 2 via audio (voice) and signalinglines. It is also coupled to the OSN though a network interface (NI) 21.In this manner, the DTMF IP 10 can interact with other components of thesystem 100, e.g., ISCP 16, via communications transmitted over OSN 34 orthrough the SSP 2. The DTMF IP 10 may be implemented using knownhardware. Accordingly, the hardware used to implement DTMF IP 10 willnot be described in detail.

The DTMF IP 10 serves as a platform by which a Centrex servicesubscriber can update his/her service information, e.g., voice dialingdirectory information, through a telephone as opposed to an Internetconnection. A Centrex service subscriber can establish a serviceupdating or management session with the DTMF IP 10, by dialing atelephone number associated with the DTMF IP 10. Dialing of the DTMF'stelephone number results in the subscriber's call being routed to SSP 2and a voice/DTMF connection to the DTMF IP being established.

DTMF IP 10 includes various security features, e.g., customeridentification and password entry requirements, as does the CICAS 32, toinsure that Centrex customers are limited to accessing and updatingthere own service records and not those of other Centrex servicesubscribers.

The second and third IPs 18 and 20 are voice dialing (VD) IPs which arededicated to supporting voice dialing services in accordance with thepresent invention. Each of the voice dialing IP's includes a networkinterface 19, 21 which couples the VD IPs to the OSN 34. The VD IP's arealso coupled to the second and third SSPs 4, 6, respectively, via voiceand signaling links. An exemplary VD IP will be discussed in greaterdetail below with regard to FIGS. 3 and 4.

The CICAS 32 will now be discussed briefly with regard to FIG. 2. Asillustrated, the CICAS 32 comprises first and second network interfaces150, 152, a processor 154 and memory 156 which are coupled together asshown in FIG. 2. The first network interface 150 links the CICAS 32 tothe Internet 30 while the second network interface 152 links the CICAS32 to the OSN 34. Thus, the CICAS 32 serves as a gateway by which aservice subscriber can gain access from the Internet, after beingauthenticated, to the various telephone system network components, e.g.,the ISCP 16, and VD IPs 18, 20. Through use of the CICAS 32 Centrexservice subscribers can manage the Centrex services to which theysubscribe via their personal computers and a Web Browser application.

The processor 154 is responsible for executing various applicationsand/or routines stored in memory to provide firewall and various servicemanagement features to Centrex subscribers. The routines executed by theprocessor 154 are stored in memory 156 until being needed by theprocessor 154, e.g. to service a Centrex subscriber. Memory 156 includesCentrex subscriber information 159. The Centrex subscriber informationincludes, for each Centrex subscriber 170, 170′, one or more subscriberidentifiers 171, 171′, the subscriber's Internet access password 172,172′, and information identifying the voice dialing IP 18, 20 assignedto store the particular Centrex subscriber's voice dialing records andto provide voice dialing services. As will be discussed below, a Centrexsubscriber's Internet access password is set by the user the first timethe subscriber accesses the CICAS 32 and identifies him or herself as anew Centrex service subscriber.

In addition to the Centrex subscriber information 159, memory 156includes a set of Centrex management server routines.

The Centrex management server routines 160 include a web access serverroutine 161, an ICAS transaction server routine 162, a database serverroutine 166, an ICAS broker routine 164, and an ICAS security routine168. The routines which form the Centrex management server application160 are responsible for providing a subscriber access to Centrexsubscriber information and hardware required to manage the subscriber'sCentrex service.

The Web access server routine 161 is responsible for initiallyinterfacing with the Centrex service subscriber and for providing webpages to be displayed. The server routine 161 interfaces with the otherroutines, which are responsible for performing security checks,retrieving subscriber data, and performing other functions.

The transaction server routine 162 is responsible for determining whatother routines need to be accessed to provide the user with a requestedservice or transaction.

The database server routine 166 is responsible for controlling access toand retrieval of locally stored information, e.g., Centrex informationincluded in the subscriber information database 159. The database serverroutine 166 interacts with the other routines 162, 164, 168 to provideCentrex subscriber information as required.

The ICAS broker routine 164 is responsible for determining theavailability of system components, e.g., system resources, to meet theneeds of the various other routines including the transaction serverroutine 162 and to retrieve data as needed from other components of thesystem 100. Accordingly, when attempting to establish a connection withan IP or another network device such as the ISCP 16, e.g., to obtaininformation, the transaction server routine 162 uses the ICAS brokerroutine 164 to determine resource availability and to arrange thedesired connection used to perform the data retrieval operation.

The ICAS security routine 168 is responsible for providing security,e.g., firewall protection. The ICAS security routine 168 is invoked whena subscriber attempts to login to manage his/her Centrex services andwhen a subscriber invokes management functions such as the passwordmodification function for which added security measures are employed. Insome embodiments security routine 168 prompts the user for additionalinformation beyond the subscriber's Internet password and then check theinformation provided by the subscriber, to add an additional level ofsecurity to certain operations such as password modification.

An exemplary voice dialing IP 18 will now be described with reference toFIG. 3. The voice dialing IP 18 can interact with the CICAS 32 to allowupdating via the Internet of a voice dialing record, sometimes called apersonal voice dialing directory, maintained for a Centrex servicesubscriber,

The voice dialing IP 18 comprises a plurality of speech recognizerarrays 70, 71 each of which is coupled by a voice line, e.g., T1 link,to an SSP. In addition to performing speech recognition functions, thespeech recognizer arrays 70, 71 serve as interfaces to the voice linesto which the IP 18 is coupled. The IP 18 also includes a switchinput/output interface 74 for coupling the IP 18 to an SSP via a controlline, e.g., over which signals in accordance with the 1129 CORE protocolmay be transmitted. In addition to the circuitry which is used tointerface with the SSP, the IP 18 includes a network interface 19 forcoupling the IP 18 to the OSN 34 and the various devices coupledthereto. Thus, the IP 18 and data included therein, can be accessed viathe switch (SSP) to which the IP 18 is coupled or by another device suchas the CICAS 32 via OSN 34.

In order to perform various operations, the IP 18 includes anapplication processor (CPU) 75 and memory 72. The memory includes a setof routines 73 and a database 85. The application processor 75 iscoupled to the switching interface 74, network interface 19, memory 72,and speech recognizer arrays 70, 71 via a local bus 79 which couples thecomponents of the voice dialing IP together.

The database 85 includes information used by the IP 18 to perform voicedialing operations as well as to create, update, and maintain personaldialing records for each Centrex subscriber served by the IP 18. Thedatabase 85 includes a set 86 of personal dialer records 88, 90, onerecord per customer. In addition it includes a set 92 of corporatedialer records 93, 94. A different corporate dialer record may exist foreach company and/or division of a company. In addition to dialingrecords which include, e.g., speech recognition models and telephonenumbers, the database includes a set of phoneme models 95, messageprompts 96, text to phoneme information 98, and a dictionary ofpotential names 99.

The application processor 75 executes the various routines 73 whencalled upon to perform various functions supported by the voice dialingIP 18. Voice dialing Web server routine 76 is responsible for presentinga Centrex voice dialing service subscriber the voice dialing informationincluded in the subscriber personal dialer record via OSN 34 and theCICAS 32, e.g., in an HTML format that can be displayed using a webbrowser. The Web server routine 76 is also responsible for receivingfrom the subscriber data representing modifications to the subscriber'svoice dialing record and for calling the Internet based personal dialerupdating routine 82 to control the updating of the subscriber's personaldialer record based on the information received from the customer viathe Internet.

The speech based personal dialer updating routine 84 is responsible forproviding a Centrex voice dialing service subscriber the opportunity toupdate his/her personal voice dialing record via DTMF and/or voice inputobtained through a conventional voice connection as opposed to anInternet connection. As will be discussed in detail below, the updatingroutine 84 uses the speech to text routine 80 to convert a spoken nameinto text information which may be used as a name entry in thesubscriber's personal dialer customer record. Personal dialer routine 78is used to actually perform voice dialing operations in contrast to thefunctions of creating and updating of personal dialer records 88, 90.

An exemplary speech recognizer array 70 will now be described in detailwith reference to Fig, 4. The speech recognizer array 70 comprises a T1interface 110, a plurality of N speech recognition circuits 104, 106,108, a plurality of N speaker independent model training circuits 105,107, 109, a DTMF tone generator/receiver circuit 102 and atext-to-speech (TTS) device 105. The components 110, 104, 105, 106, 107,108, 109, 102, 103 of the speech recognizer array 70 are coupledtogether by a high bandwidth, e.g., voice, bus 101 and a lowerbandwidth, e.g., data/control, bus 103. The data/control bus 103 iscoupled to application processor bus 79. A CPU 116 is provided forcontrolling allocation of speech recognition resources, e.g., assignmentof the speech recognition circuits 104, 106, 108, to service differentcalls and to oversee bus traffic as well as interaction with the IP'sapplication processor 75. T1 interface 110 acts as the interface betweenthe voice connection with the SSP and the speech recognition circuits104, 106, 108, SI model training circuits 105, 107, 109, DTMF tonegenerator/receiver circuit 102, and TTS device 103 of the speechrecognizer array 70.

During a voice model training operation, one of the speech recognitioncircuits 104, 106, 108 is used to identity a spoken name in one or morespeech samples received from a subscriber while the corresponding SImodel training circuit 105, 107, 109 is used to generate a speakerindependent speech recognition model, e.g., a phonetic representation ofthe spoken name.

The speech recognition circuits 104, 106, 108 are used during voicedialing operations to detect names in a voice dialing directory, spokennumbers, and/or commands. Speech recognition circuits 104, 106, 108normally perform speaker independent speech recognition operations.However, in some embodiments, they are also used to perform speakerdependent recognition of one or more names.

During voice based model training operations, the speech recognitioncircuit 104, 106, 108 process the received speech corresponding to aname and compares it to a wide range of name hypothesis corresponding toan extensive dictionary 99 of possible names. In this manner, a spokenname can be recognized and a text version of the name can be generated,e.g., from information included in the dictionary 99 of possible names.A SI speech recognition model corresponding to the name is generated byone of SI model training circuits 105, 107, 109. In this manner, aphonetically accurate model of the name is generated for future speechrecognition purposes. As will be discussed below, the text version ofthe recognized name and corresponding generated speaker independentspeech recognition model may be stored in a subscriber's voice dialingcustomer record for use during a subsequent voice dialing operation.

During voice dialing operations, speech recognition models, e.g.,speaker independent speech recognition models, stored in a subscriber'svoice dialing customer record 88, 90 are retrieved from the database 85and supplied to speech recognition circuit 104, 106 or 108 assigned toservice the call. The retrieved models may be used in conjunction withspeaker independent models of commands, names of locations, and/ornumbers. The speaker independent speech recognition models obtained fromthe customer record 88, 90, which may be in the form of phoneticrepresentations of the modeled names, are used to determine which, ifany, name in a subscriber's personal voice dialing directory was spoken.The other speaker independent models are used during a voice dialingoperation to detect spoken commands, names of locations and/or numbers,e.g., digits to dial or numbers spoken in response to a prompt forinput.

In one embodiment, during a voice dialing operation, recognition ofphones, or phonemes which maybe used synonymously, is performed in anintegrated way as part of the overall speech recognition operation. Aspart of the speech recognition operation, the incoming speech, afterappropriate processing, is compared against a network of possiblephone/phoneme sequences dictated by the speech recognition models usedand associated grammar. The best path in the network, corresponding to aparticular word, phrase or result, e.g., no valid word recognized, isselected during the speech recognition operation and corresponding word,phrase or result is declared as the outcome of the speech recognitionoperation. Accordingly, in at least one exemplary voice dialingimplementation, phones are recognized in the context of the expectedphone-strings, dictated by a grammar that defines the space of possibletext strings, to be spoken by the caller, for that application, e.g.,voice dialing.

When performing a voice dialing operation, in the event that 1) a namefrom the subscriber's personal dialing directory is not recognized, and2) a telephone number is not detected in the form of a spoken number,DTMF input, or a combination of spoken numbers and DTMF input, thespeech recognition circuit 104, 106 or 108 may, depending on how thesubscriber has configured his personal dialing record, use one or morespeech recognition models obtained from the corporate dialer records 92in an attempt to identify a name in received speech.

DTMF tone generator/receiver 102 is responsible for detecting DTMFsignals and generating DTMF signals when needed, e.g., to cause the SSPto route a call to a destination telephone number. By using the speechrecognition circuit 104, 106 or 108 in combination with DTMF tonegenerator/receiver 102 a voice dialing subscriber can perform a dialingoperation by supplying a telephone number to be dialed as a combinationof spoken numbers and numbers input by depressing telephone keys whichcauses DTMF input signals to be generated.

The TTS device 105 is included in the speech recognizer array 101 sothat messages and prompts 96 can be played to a subscriber as required.Prompts may be played, e.g., to report recognition problems, to play anannouncement of the recognized name to be dialed, and to prompt forinput when needed.

The components of the speech recognizer arrays 70, 71 operate inconjunction with the application processor 75 and various routines 72 toprovide voice dialing services and the ability to update and revisesubscriber's voice dialing directories through the use of audio and DTMFinput. In the case of Internet based updating of a subscriber's voicedialing records, speech recognizer arrays 70, 71 are not used. This isbecause, during the Internet based voice dialing record updatingprocess, there is no audio connection via the SSP to the subscriber.

Having described the components of the communications system 100,initial provisioning of a service to a Centrex subscriber using thevarious system components will now be discussed with reference to FIGS.5-9. Subsequent management of a Centrex service via the Internet willthen be described with regard to FIG. 10. The top row in FIGS. 5-9illustrate the various system components involved in performing thesteps illustrated in the figure. The starting, e.g., initial condition,is shown in the block immediately below the row of system elements. Theexit condition of the system 100, at the end of the processingillustrated in each figure, is shown in the last block located at thebottom of FIGS. 5-99. In FIGS. 5-9 arrows are used to show the transferof data, voice, and/or control signals between various components of thesystem 100. Labeled boxes within arrows are used to represent processingoperations or steps.

FIG. 5 illustrates the initial processing associated with provisioningCentrex service for a new subscriber. For purposes of this example, itis assumed that the first subscriber, located at customer premises 22,seeks to set up Centrex service for the first time. Accordingly, at thestart of FIG. 5, the initial conditions <A_(—)00b> are that the system100 is in operation and that the customer premise equipment, e.g.,telephone 38, is linked to any one of the SSPs in the system 100.

In step Al the person wishing to subscribe to Centrex service places acall to a Centrex service subscription number. The call is received bythe SSP 4 to which the caller is coupled. Thus, in the example thesecond SSP 4 becomes the originating SSP. In step A2, the originatingSSP 4 sends an initial address message (IAM) to the SSP 2 correspondingto the dialed Centrex service subscription number. Thus, the SSP towhich the Centrex customer service representative (CCSR) 44 is connectedbecomes the terminating SSP for this portion of the processing.

In step A3, the service representative responds to the caller andprompts the caller for subscription information including the landlineand mobile telephone numbers for which Centrex service is to beprovided. Then in step A4, the caller provides subscription information,including the requested telephone numbers, to the CCSR 44. In responseto receiving the requested Centrex subscription information, the CCSR44, in step AS, provides Centrex-service activation instructions to thecaller. The activation instructions include a temporary PIN valuecorresponding to the last for digits of the caller's landline phone towhich Centrex service is to be added, an 800 number to call from thecaller's landline phone to activate the Centrex service, and informationon how soon the caller can call to activate the subscribed services.

The condition existing upon completion of the steps illustrated in FIG.5 are shown in exist state A_(—)00e. In exist state <A_(—)00e>, the CCSR44 has the new Centrex subscriber account information and the servicesordered are not yet available for use by the new Centrex subscriber.

The steps associated with the creation of a new customer's callprocessing record and the initial setting of AIN triggers on the newsubscriber's telephone lines are shown in FIG. 6. In FIG. 6, the initialstate <B_(—)00b> is the same as the FIG. 5 exit state <A_(—)00e>.Accordingly, in state <B_(—)00b> the CCSR 44 has the new Centrexsubscriber account information and the services ordered are notavailable for the new subscriber's use. The process illustrated in FIG.6 begins with step B1 wherein the CCSR 44 initiates a service creationwork order via entering the new subscriber's subscription informationinto operator terminal (OT) 49 and having that information supplied tothe service order processing circuitry 48. The SOP circuitry 48 createsfrom the subscription information a service order which, in step B2, issupplied to the AIN provisioning system circuitry 46.

In response to the service order, the APS circuitry creates a callprocessing record (CPR) for the new subscriber which includes a set ofcall processing instructions designed to control the SSPs and othernetwork elements, in response to messages initiated by AIN triggers seton the subscriber's lines. In step B3, the APS 46 forwards the newsubscriber's CPR, including the subscribers assigned temporary PINvalue, to the MSAP 69 for storage and future use.

In step B4, the MSAP 69 acknowledges to the APS circuitry 46 receipt andstorage of the new subscriber's CPR in the MSAP's set of customer data71. Next, in step B5, the APS circuitry 46 updates the SSPs associatedwith the new subscriber's telephone line by setting one or more AINtriggers on the subscriber's lines, e.g., terminating attempt triggers.In the FIG. 1 example, when configuring the account for the firstcustomer located at CP1 22, a TAT trigger would be set at the second SSP4 and at the MTSO servicing the first subscriber's mobile phone 37.

At the end of the process illustrated in FIG. 6, in exit state<B_(—)00e>, it can be seen that one or me Centrex services areassociated with the new subscriber's lines and that the CPR for the newsubscriber includes the subscriber's temporary PIN value.

FIG. 7 illustrates the process by which a new Centrex subscriberactivates his/her Centrex service by dialing a number corresponding tothe Centrex DTMF IP 10, for the first time. In FIG. 7, the initial state<C-00b> is the same as the FIG. 6 exit state <B_(—)00e>. Accordingly, instate <C-00b> one or more Centrex services have already been associatedwith the new subscriber's lines and the new subscriber's CPR include: atemporary PIN value.

To activate the Centrex service, in step C1, the user places a call fromthe subscribing line, e.g., phone 38, to the Centrex service activationnumber previously provided. Since the dialed Centrex activationtelephone number corresponds to Centrex DTMF IP 10, the SSP 4 which, inthis example is the originating SSP, sends, in step C2, an initialaddress message (IAM) to SSP 2 to which DTMF IP 10 is connected. Thus,in this example, SSP 2 is the terminating SSP. In step C3, terminationSSP 2 then forwards the new subscriber's call to the Centrex DTMF IP 10to which the call was directed.

With the voice connection between the DTMF IP and the new subscriberestablished, the DTMF IP 10 plays, in step C4, PIN change instructionsto the new subscriber. Then, in step C5, the user enters, in the form ofa DTMF input or spoken digits, a new PIN value. The DTMF IP 10, whichincludes speech recognition capability, collects the new PIN value fromthe subscriber and, in step C6 forwards the new PIN value to the ISCP 16via a message sent using the 1129+ (SR3511) protocol.

In step C7, the ISCP 16 updates the new subscriber's PIN informationstored in the customer database 71 and sends a message to the DTMF IP 10acknowledging receipt of the PIN information and the updating of the newsubscriber's CPR. In response to the message from the ISCP 16, the IP10, in step C8 plays one or more service-activation confirmationmessages to the subscriber indicating that the subscriber's Centrexservice has been successfully activated.

Accordingly, at the end of the process illustrated in FIG. 7 thecondition at the exist state <C-00e> is that the new subscriber's CPRincludes a valid PIN value that was provided by the subscriber, e.g., 4digits or letters that can be entered using a standard telephone keypad.

FIG. 8 illustrates the process associated with a new Centrex subscriberwho has already activated his/her service via the telephone, connectingvia the Internet to manage his/her Centrex services for the first time.In FIG. 8, the initial state <D-00b> is the same as the FIG. 7 exitstate <C_(—)00e>. Accordingly, in state <D-00b> the new subscriber's CPRincludes a valid user provided PIN value.

The process illustrated in FIG. 8 begins in step D1 wherein thesubscriber submits, using a Web Browser application, a request to theCICAS 32 for access to a Centrex Web access default web page. Therequest is forwarded through the network interface 150 to the Web accessserver routine 161. Next, in step D2, the Web access server routinereturns to the subscriber a Web Access default page 100, e.g., a page ofthe type illustrated in FIG. 10. The subscriber's Web Browser displaysthe default page on the screen of computer 36.

Next, the subscriber requests to manage his/her existing Centrex accountby, e.g., clicking on the displayed “MANAGE YOUR EXISTING CENTREXACCOUNT” option. This causes, in step D3, the subscriber's Web browserapplication to send a Centrex service access request to the Web accessserver routine 161. In response to the service access request, in stepD4, the Web access server routine 161 initiates a secure communicationssession with the subscriber's Web browser and then sends the subscribera request for the subscriber's CallingPartylD and PIN value. The requestmay be presented to the use in the form of a data entry screen 1100 asshown in FIG. 11.

In response to the request, in step D5, the subscriber provides to theWeb access server routine 161, his/her CallingPartylD, e.g., Centrextelephone number, and PIN.

Following the receipt of the information, in step D6, the Web accessserver routine 161 requests that the ICAS security routine 168authenticate the user based on the received information and ensure thatthis is a new account. Then, in step D7, the ICAS security routinerequests that the ICAS transaction server retrieve the current PIN valuestored in the subscriber's CPR. In step D8, the ICAS transaction serverroutine requests a copy of the subscriber's current CPR from the ICASbroker routine 164 which is responsible for performing the actualretrieval operation.

In step D9 the ICAS broker routine requests the subscriber's CPR fromthe ISCP 16. In step D10, the ISCP 16 sends the subscriber's CPR to theICAS Broker routine 164. The Broker routine 164 then passes, in stepD11, the retrieved CPR to the ICAS transaction server routine 162. Instep D12 server routine 162 forwards the PIN, obtained from the CPR, tothe ICAS security routine 168. In step D13, the ICAS security routine168 compares the recently provided PIN to the retrieved PIN obtainedfrom the CPR and, assuming they match, sends an authentication messageto the Web access server routine 161.

In step D14, the Web access server routine requests that the subscriberenter a long, e.g., 8 digit, alphanumeric password to by used for futureInternet access operations. The server routine 161 also requests thatthe subscriber complete an initial log-in form which providesinformation used to implement the subscriber's Centrex services. In stepD15, the user enters a long password, completes the initial log-in formand repeats the password entry to confirm the first value entered. Instep D16, the Web access server 161 ensures that the password meetsnecessary requirements and that any mandatory fields are completed onthe initial log-in form in step D16. The login password and log-in forminformation are passed to the ICAS security server routine 168.

In step D17, the ICAS security server routine 168 saves the submittedlong password and other service information, e.g. in the local set ofCentrex subscriber information 159, and then signals the subscriber thatthe set-up operation has been successfully completed.

Accordingly, at the end of the process illustrated in FIG. 8, the newsubscriber is authenticated, a long password to be used for Internetaccess to Centrex services exists for the subscriber and one or moreCentrex services are authorized for the subscriber.

The process of accessing and updating Centrex subscriber accountinformation via the Internet 30, after an initial Internet accessoperation, is illustrated in FIG. 9. In FIG. 9, the initial state<E-00b> is the same as the FIG. 8 exit state <D_(—)00e>. Accordingly, instate <E-00b> the subscriber's long PIN used for Internet access andother account information has already been stored in the set of Centrexsubscriber information 159 maintained by the CICAS 32.

In step E1, a customer request to access his/her Centrex account isreceived by the Web access server routine via the Internet 30 and thenetwork interface 150. In step E2, the Web Access Server Routine returnsthe default web page 1000 to the subscriber. In step E3, the subscriberrequests to manage has/her existing Centrex account. Then, in step E4,the Web access server routine 161 initiates a secure session with thesubscriber's Web browser application and requests Calling Party Id andPIN (password) information, e.g., by sending the page 1100 to thesubscriber's browser.

In step E5, the user provides the requested ID and password information.Then in step E6, Web access server routine 161 requests that the ICASsecurity routine 168 authenticate the user based on the received CallingParty Id and PIN. In step E7, the ICAS security routine 168authenticates the subscriber by comparing the received ID and PINinformation to account information, e.g., PIN information for theidentified subscriber, stored in the ICAS's set of Centrex subscriberinformation 159 and, assuming the PIN matches the stored PIN, authorizesthe Web access server routine to provide the subscriber access tohis/her account information.

In step E8, assuming the subscriber was authenticated, the Web accessserver routine 161 presents the subscriber with a Centrex managementservice selection page which lists options for managing the variousservices to which the user subscribes. FIG. 12 illustrates an exemplaryCentrex management service selection page 1200 wherein the subscriber ispresented with the option of managing his/her voice dialing service,call forwarding service, call screening service, account/passwordinformation, or voice mail.

In step E9, the user communicates the selected management operation tothe Web access server routine 161. Then in step E9, the Web accessserver retrieves the Centrex subscriber information it has available toit from its own database relating to the selected service and, ifanother device e.g., IP is responsible for maintaining customer recordsrelating to the selected service, couples the subscriber to the deviceresponsible device. The subscriber is also presented in step E10 with anew web page reflecting the retrieved account information relating tothe service to be managed.

Consider for example, if the first subscriber had in step E9 selected tomanage his/her voice dialing service. In such a case, in step E10, thesubscriber would be coupled via the Internet, ICAS 32 and OSN 34 to VDIP 18 where the subscriber's voice dialing records reside. In addition,the subscriber would be presented with a voice dialing management pagesuch as the page 1300 illustrated in FIG. 13 showing the existing voicedialing information.

At the end of the process illustrated in FIG. 9, the subscriber has beenauthenticated and has been provided access to one or more Centrexservices, e.g., account management services. This condition is shown inthe FIG. 9 exist state <E_(—)00e>.

Internet based management of a subscriber's Centrex voice dialingrecord, maintained in the VD IP 18, will now be discussed. At the end ofthe process shown in FIG. 13, in the case of a voice dialing directorymanagement operation, the user will be connected to the VD IP 18 via theInternet. The VD IP's voice dialing web server routine 76 interfacesbetween the subscriber's web browser and the various routines of thevoice dialing IP. The Internet Based personal dialer updating routine 82is activated by the voice dialing web server routine in response to thesubscriber's request to manage/update his or her personal voice dialingservice.

The routine 82 begins by displaying via the user's Web Browser, VoiceDialing window 1300 illustrated in FIG. 13. The window displays voicedialing information included in the subscriber's voice dialing recordwhich the user may wish to modify, add to, or delete. It also displayscontrol options, e.g., options such as the option of adding a new entry1308, and modifying the subscriber's password, which the user canselect, e.g. by double clicking on the displayed option.

The voice dialer information displayed to the subscriber includes, e.g.,the subscriber's Centrex phone number 1302, the subscriber's mobilephone number 1304, the subscriber's easy access number 1306, the name ofa corporate dialer 1320 if the subscriber has designated one to be usedin conjunction with the subscriber's personal dialer, and voice dialingentries, e.g., calling records 1310, 1312, 1314, 1316. Each callingrecord corresponds to one individual or party who the voice dialingsubscriber can call through the use of voice dialing, e.g., by speakingthe person's name or nickname.

Unlike some known voice dialing systems, the voice dialing system of thepresent invention allows multiple, e.g., different, names correspondingto the same party to be used in initiating a voice dialing call. In theillustrated embodiment, both a primary name and a nickname aresupported. Accordingly, multiple, e.g., alternative names for the sameparty or individual, can be stored and associated with the sametelephone number or numbers in a calling record.

The primary name and nickname form a set of names with which one or moretelephone numbers are associated. As will be discussed below, a speakerindependent speech recognition model is generated for each name andnickname in a set of names and stored as part of a subscriber's voicedialing record.

One or more telephone numbers, corresponding to different locations ordifferent individuals such as a secretary, are associated with each setof names. During a voice dialing operation, the subscriber can call aparty by stating the party's name and the desired location thesubscriber wishes to call. While the term location is used here,location is used a telephone number identifier. Thus, in the context ofthe present application, the “location” may actually be an identifier ofa person such as a secretary associated with the named party orindividual. If the VD IP detects a name included in the subscriber'svoice dialing record and detects a spoken location, the telephone numberin the record corresponding to the name and location will be used. Ifonly a name is detected, it is assumed that the first telephone numbercorresponding to the name is to be used for dialing purposes.

Locations which may be specified can be selected from the following listof identifiers:

OFFICE OFFICE 1 Office 2 Office 3 OFFICE 4 Second Office Branch OfficeHeadquarters Secretary Home Home 1 Residence Residence 1 Residence 2Residence 3 Mobile Mobile 1 Mobile 2 Mobile 3 Temporary Location

By limiting the list of locations which can be designated, the need forthe VD IP to store speech recognition models for custom locations isavoided and a single set of speaker independent speech recognitionmodels can be used for detecting the spoken name of a location formultiple subscribers. For example, to support the above list of 20possible locations, 20 speaker independent speech recognition models arestored by the VD IP 18 and multiple subscribers.

When presented with the voice dialer information 1300, the subscribercan choose to modify the displayed information, add a new entry or tochange his/her password, e.g. by double clicking on an existing entry,icon 1308, or icon 1309, respectively. A subscriber's selections arecommunicated by the subscriber's computer to the VD IP 18 via theInternet.

If the subscriber selects to add a new entry or to edit an existingentry, the subscriber is presented with a voice dialing entrymodification page 1400 which is displayed by his/her browser. The page1400 includes information relating to the entry to be added or modified.

While having much of the same entry information as the voice dialer page1300, the voice dialing entry modification page includes additionalinformation relating to an entry as well as entry modificationinformation, e.g., an icon 1407 which may be selected for display alisted location names from which the subscriber can choose when editingthe contents of an entry. In addition, the page 1400 includes a submitupdated entry icon 1410. The icon 1410 may be selected by the user,e.g., by double clicking, to submit the entry information to the VD IP18 after the user makes the desired changes to the entry. In response toselection of the submit updated entry icon 1410, the user's web browsertransmits the entry information, as modified by the user, to the VD IP18.

In FIG. 14, the first column 1401, of entry 1312, is a name entry. Inthe second column 1402, model type information is displayed. Asdiscussed above, speech recognition models used to recognize names canbe generated from speech or text in accordance with the presentinvention. To identify how the speech recognition model was generatedmodel type information is maintained. An S is included in column 1402 ifthe model associated with the name was generated from speech while a Tis included in column 1402 if the speech recognition model was generatedfrom text, e.g., the text in column 1401.

The third column 1403 lists the nickname, if any, associated with thesame party or individual, whose name is listed in the first column 1401.The fourth column, 1404 identifies how the speech recognition model wasgenerated for the nickname listed in column 1403, e.g., whether it wasgenerated from speech or text.

The subscriber can change an S to a T in columns 1402, 1404 if desired.This will have the effect of causing the speech recognition modelassociated with the corresponding name to be generated from the text ofthe name instead of the speech previously provided by the user.Generally, names trained from a user's speech will be more reliable thanthose generated from text. Accordingly, the changing from a speechrecognition model generated from the user's speech, to one generatedfrom text, is expected to incur infrequently.

The fifth column 1405 illustrated in FIG. 14 lists location informationwith which a telephone number for the named individual or party isassociated. For each location listed in column 1405, a correspondingtelephone number is listed in the sixth column 1406.

While places for five possible locations and telephone numbers are shownin FIG. 14, it is to be understood that any number of differentlocations can be supported with, e.g., a different telephone numberassociated with each location.

In accordance with one embodiment, in the absence of the subscriberexplicitly designating a telephone number in a calling entry as thedefault telephone number, the first listed telephone number is used as adefault telephone number for the calling entry 1400. A subscriber maydesignate one of the telephone numbers in the calling entry 1400 as adefault telephone number by double clicking on the box in column 1407located to the right of the telephone number. A “D” in the boxassociated with the telephone number indicates that it is selected asthe voice dialing default for a given calling entry 1400. In FIG. 14, itcan be seen that the mobile telephone number 301-260-7465 has beenselected as the default telephone number to be associated with the name“JOHN DOE” and the nick-name “JOHHNNY D”. In such an embodiment, duringa voice dialing operation, when a name or nickname is detected in speechbut a spoken location is not detected, the default telephone numberassociated with the detected name or nick-name is used to route thecall.

The subscriber can edit the information shown in the columns 1401, 1402,1403, 1404, 1405, 1406 as desired. Upon completing the modifications tothe entry 1312, the user selects the submit updated entry icon 1410 andthe revised entry information is transmitted via the Internet to the VDIP 18. As will be discussed in detail below, the VD IP 18 updated thecalling record corresponding to the entry as may be required by theuser's changes.

While an existing entry 1312 is shown in FIG. 14, a new entry would bedisplayed in a similar fashion with the information in each of the sixcolumns being blank. The user need not fill in columns 1402, 1404 sincespeech recognition models for new entries created via the Internetupdating process are generated from the text name with columns 1402 and1404 then being updated VD IP 18 accordingly. The speech recognitionmodel generated from the supplied text name will later be replaced witha voice trained model should recognition accuracy achieved with thespeech recognition model generated from text prove unsatisfactory to thesubscriber.

FIG. 15 illustrates an exemplary voice dialing customer record 1500which is stored by the VD IP 18 for a voice dialing subscriber. Therecord 1500 may be used as any one of the customer records 88, 90illustrated in FIG. 3. One voice dialing record 1500 is stored for eachsubscriber. As illustrated, the record comprises much of the sameinformation that was shown in FIGS. 13 and 14. A ′ mark is added to eachof the numbers in FIG. 15, which corresponds to previously discusseddata shown in figs. 13 and 14 to distinguish between the stored data inthe record 1500 and the displayed data of FIGS. 13 and 14.

Note that in addition to the data previously discussed with regard toFIGS. 13 and 14, the customer record 1500 includes the subscriber'sInternet PIN 1501 and the speech recognition models, SI MODELs 1-7,shown in columns 1502 and 1504. In the illustrated embodiment, one modelspeaker independent speech recognition model is stored for each name andnickname included in the customer's record.

As discussed above, speech recognition models may be generated from aspeech sample provided by the subscriber, e.g., over a voice telephoneconnection, or from text provided via the Internet. When dialing entriesare initially generated from speech, phoneme to text conversion is usedto generate a text name corresponding to a generated speech recognitionmodel. In addition, the model type identifier, MT, for the generatedmodel, is set to S indicating the model was generated from speech. Oftenthe text spelling of the name may include errors resulting from thephoneme to text conversion process. In accordance with the presentinvention, a subscriber can update the text name, via the Internet,without affecting the corresponding speech model. This can be done bymodifying the text of the name but leaving the model type identifier setto S. In accordance with the present invention, this results in the textname information being updated, e.g., corrected, with the speechrecognition model being left unaltered. Should a subscriber wish toreplace a speech recognition model generated from speech with one fromtext, the subscriber can change the “S” in the model type field to a“T”. This will result in a speech recognition model being generated fromthe text of the name. The generated model will be stored in place of thepreviously generated speech recognition model. Thus, when the text of aname is altered, the text entry is updated. However, when there is anexiting speech recognition model which was generated from speech (MT=S)it is replaced only if the user modifies the model type identifier toindicate the model is to be generated from text.

FIG. 16 illustrates voice dialing entry updating sub-routine 1600 whichis part of the Internet based personal dialer updating routine 82. Theentry updating sub-routine 1600 begins in step 1602 when it is executedby the VD IP 18 in response to receiving updated dialing entryinformation. From start step 1602 operation proceeds to step 1604,wherein the received text information, e.g., name, nickname, and/orlocation information, as well as the received number information, e.g.,telephone number information, is stored in the portions of the customerrecord to which the received information corresponds. In this manner,old or missing information is updated with the latest version of theinformation which the subscriber has submitted.

From step 1604, operation proceeds to step 1606 wherein a determinationis made as to whether the model type identifier, MT, associated with thename, is equal to S. That is, in step 1604 a determination is made as towhether a speech recognition model corresponding to the name, hasalready been generated from speech (as indicated by the presence of theS in the model type field).

If an S exists in the MT field it indicates that the existing speechrecognition model is not to be altered and operation proceeds to step1616.

However, if in step 606, it is determined that the MT field does notequal S, operation proceeds to step 1608. In step 1608, a determinationis made as to whether the value in the MT field was changed from S to Tindicating that a speech recognition model should now be generated fromthe text of the name. If the model type field value was changed from Sto T operation proceeds to step 1612. Otherwise operation proceeds tostep 1610.

In step 1610, a determination is made as to whether the name wasmodified by the subscriber. This check can be made by comparing theprevious value in the name field, i.e., the value supplied to thesubscriber's web browser for display, to the value supplied by the webbrowser to the VD IP 18 as part of the modified entry information. Ifthe name was not modified, updating to the speech recognition model isnot required and operation proceeds directly to step 1616. If, however,in step 1610 it is determined that the name was modified, operationproceeds to step 1612.

In step 1612, a speech recognition model is generated from the text ofthe name, obtained from the subscriber via the Internet. A text tophoneme operation is normally performed as part of the speakerindependent model generation process. Text to phoneme information 98 maybe used when converting the text information to phoneme information.

After the speaker independent speech recognition model is generated instep 1612, operation proceeds to step 1614 wherein the generated SIspeech recognition model is stored in the customer record with the modelbeing associated with the name from which it was generated. As part ofthe storing operation, a model type value associated with the generatedmodel is set to T if it is not already set to that value, e.g., becausea model did not previously exist. Operation proceeds from step 1614 tostep 1616.

Steps 1616 though 1624 deal with the updating of the speaker independentspeech recognition model corresponding to a nickname. The updatingprocesses is generally the same as that described above with regard tothe updating of the speaker independent speech recognition modelcorresponding to the name.

In step 1616, a determination is made as to whether or not the MT valuecorresponding to the nickname is set to S. If the MT value is set to S,operation proceeds to step 1626 with the SI speech recognition modelcorresponding to the nickname being left unaltered.

However, if in step 1616, if it is determined that the MT field does notequal S, operation proceeds to step 1618. In step 1618, a determinationis made as to whether the value in the MT field was changed from S to Tindicating that a speech recognition model should now be generated fromthe current text of the nickname. If the model type field value waschanged from S to T operation proceeds to step 1622. Otherwise operationproceeds to step 1620.

In step 1620, a determination is made as to whether the name wasmodified by the subscriber. This check can be made by comparing theprevious value in the name field, i.e., the value supplied to thesubscriber's web browser for display, to the value supplied by the webbrowser to the VD IP 18 as part of the modified entry information. Ifthe name was not modified, updating of the speech recognition model isnot required and operation proceeds directly to step 1626. If, however,in step 1620 it is determined that the nickname was modified, operationproceeds to step 1622.

In step 1622, a speech recognition model is generated from the text ofthe nickname obtained from the subscriber via the Internet.

After the speaker independent speech recognition model is generated instep 1622, operation proceeds to step 1624 wherein the generated SIspeech recognition model is stored in the customer record with the modelbeing associated with the nickname from which it was generated. As partof the storing operation, a model type value associated with thegenerated model is set to T if it is not already set to that value,e.g., because a model did not previously exist. Operation proceeds fromstep 1624 to step 1626.

In step 1626, the entry updating sub-routine is stopped pending itsre-execution to processes another sec of modified voice dialing entryinformation.

As discussed above, a voice dialing service subscriber can also updatehis/her voice dialing record through the use of an ordinary telephone.In such a case, the subscriber calls a telephone number assigned toservicing voice dialing customers. The subscriber is first connected tothe Centrex DTMF IP 10 which collects the user's ID, e.g., Centrextelephone number and PIN. After the ISCP 16 validates the subscriber,assuming the subscriber's PIN is correct, the subscriber is connected tothe VD IP 18. The phoneme based personal dialer updating routine 84 isinvoked, and the subscriber's voice dialing record is retrieved based onthe User ID (Centrex telephone number) obtained using ANI information orinformation provided by the subscriber. The updating routine 84 promptsthe subscriber to indicate, e.g., by stating “1” to create a new voicedialing entry or “2” to update an existing voice dialing entry what thesubscriber wishes to do. The subscriber may be presented , with otheroptions as well.

If the VD IP 18 detects a “1” response either by the use of speechrecognition or DTMF detection, the subscriber is prompted for the nameof the party to be added to the voice dialing directory. At this pointthe subscriber states the name and the audio obtained from thesubscriber is captured for further processing. The subscriber is alsoprompted for the telephone number of the party to be added. Locationinformation may also be obtained from the subscriber if multipletelephone numbers are to be entered for the party. Telephone numberinformation is normally entered via DTMF input generated by pressingkeys on a telephone keypad. Alternatively the number information can beentered using spoken numbers.

In the case when the VD IP 18 detects a “2” in response to its initialprompt, indicating that an existing calling entry is to be modified, theVD IP 18 prompts the subscriber for the name of the party whose entry isto be modified. At this point the subscriber states the name and thespeech obtained from the subscriber is captured for further processing.The subscriber is also prompted for location and telephone numberinformation. The provided telephone number and location information isused to update the telephone number information included in the entrywhich the speech corresponding to the name is used to create a speechrecognition model for the name.

FIG. 17 illustrates an audio based SI model updating sub-routine 1700which is part of the speech based personal dialer updating routine 84.The sub-routine 1700 is executed, during the entry creation or updatingprocess, once an audio sample of a name has been collected. Thesub-routine 1700 starts in step 1702 and proceeds to step 1706. In step1706 the speech recognizer 104, 106 or 108 is used to process the audiosample 1704 of the name. The dictionary of potential names 99 whichincludes phonetic information as well as text information may be used bythe speech recognizer during this process. In step 1708 a speakerindependent model training circuit 105, 107, or 109 is used to generatea speaker independent speech recognition model of the name. The SIspeech recognition model may take the form of a phonetic representationof the modeled name. Optionally, a speaker dependent speech recognitiontraining circuit may be used to generate a speaker dependent speechmodel instead of a speaker independent model. The generated speechrecognition model is stored in step 1710 in the entry being created orupdated. As part of the storage process, the model type identifier MT,associated with the speech recognition model, is set to “S” indicatingthat the SI speech recognition model was generated from speech. Oncestored, a determination is made in step 1712 as to whether or not a textname entry already exists for the name which was just modeled. In thecase where an existing entry is being updated the answer to the inquiryin step 1712 will be yes. In such a case operation proceeds to step1718.

However, if in step 1712 it is determined that a text name entrycorresponding to the generated model does not already exist, operationproceeds to step 1714. In step 1714, a text entry for the recognizedname is generated, e.g., from information in the dictionary 99.

Then in step 1716, the generated text name, e.g., text entry, is storedin the subscriber's voice dialing record in a manner that associates thetext name with the generated SI speech recognition model. From step 1716operation proceeds to step 1718 wherein the updating sub-routine 1700 isstopped.

In the above described manner, a new voice dialing entry can be createdfrom a spoken name with the text name and the speech recognition modelincluded in the entry being generated from the same audio sample, e.g.,through the use of speech recognition and speech recognition modeltraining techniques, respectively. In addition, speech recognitionmodels of names which were originally generated from text, can bereplaced with speech recognition models generated from spokenutterances.

Accordingly, subscriber's can take advantage of the ease of creatingvoice dialing entries via the Internet and the use of speech recognitionmodels generated from text, while still being able to train speechrecognition models via audio samples in cases where the recognitionmodels generated from text provide unsatisfactory recognition results.

Having discussed the creation and maintenance of a subscriber's voicedialing record in detail, use of the record and the voice dialing IP 18to place one or more telephone calls will now be discussed.

FIGS. 18-20 are call flow diagrams. The diagrams illustrate processedsteps and the transfer of messages, data, and other signals betweencomponents of the system 100, which occur as part of various Centrexsupported voice dialing operations. The top row in each of FIGS. 18-20illustrate system components which are involved in placing a call usingthe voice dialing service of the present invention. In FIGS. 18-20 boxesin rows, other than the top row of FIGS. 15-18, represent steps whilearrows are used in FIGS. 15-18 to represent the transfer of signals,data, and/or messages between system components.

FIG. 18 illustrates a basic voice dialing call implemented from aCentrex subscriber's telephone line which has been provisioned for voicedialing service. For purposes of explanation, it will be assumed thatthe call is placed by the first subscriber from the first subscriberpremises 22 using landline phone 38.

The process of placing a call illustrated in FIG. 18, begins in step201, with the first subscriber picking up the telephone 38 and dialing*0, indicating a request for a voice dialing operation. As part of theprovisioning of voice dialing service for the first subscriber, a CDP(customized dialing plan) feature code trigger was set at the SSP 4 towhich the first subscriber's phone is connected. While a CDP featurecode trigger is used for Centrex subscribers, a vertical access codetrigger is used to provide voice dialing services to non-Centrexsubscribers, e.g., ordinary residential telephone customers. Forpurposes of explanation, voice dialing will be explained assuming use ofa CDP feature code trigger with the understanding that the same orsimilar processing would occur in response to activation of a verticalaccess code trigger, used to provide non-Centrex subscribers with voicedialing service in accordance with the present invention.

In step 202, the CDP feature code trigger set at SSP 4 on the firstsubscriber's line is hit and call processing is paused. Then, inresponse to activation of the trigger, the SSP 4 launches, in step 203,a query for call processing instructions to the ISCP 16. The queryincludes a user ID corresponding to the first subscriber.

In step 204, the ISCP 16 uses service key analysis on the USER ID,included in the received query, and opens the CPR corresponding to theindicated user, i.e., the first subscriber. In accordance with thepresent invention, the USER ID may be, e.g., the subscriber's Centrextelephone number. Since the feature code trigger *0 triggered the query,the ISCP 16 knows from the query that the subscriber is trying toinitiate a voice dialing operation. The opened CPR instructs the ISCP tosend a Send To Outside Resource (STOR) command to the SSP instructingthe SSP to send the call to a specific IP address, where the address isthat of the voice dialing IP 18 which services the first subscriber. Theaddress of the voice dialing IP 18 assigned to service a particularcustomer is stored as part of the first subscribers CPR.

In step 205, the STOR message with the VD IP address is transmitted tothe SSP 4. Included in the STOR message is a play PROMPT 1 instruction.Where PROMPT 1 is an audio prompt which requests the name to be dialed.

In response to the STOR message, in step 206, the SSP 4 sets up an ISUPconnection to the VD IP 18 specified by the received IP address andpasses the IP the play PROMPT 1 instruction. At this point the caller isnow connected to the voice dialing IP 18. The VD IP may be local orremote to the SSP 4, i.e., the SSP 4 may use a voice dialing platformcoupled to a different SSP.

In step 207, the VD IP 18 recognizes, e.g., based on the USER ID, thesubscriber just connected to the VD IP 18 and retrieves their voicedialing information, e.g., personal dialing directory informationincluding speaker independent speech recognition models and telephonenumbers corresponding to modeled names. Once the IP 18 is ready toperform a voice dialing operation, as part of step 207, the VD IP 18retrieves PROMPT 1 from its memory and plays PROMPT 1. PROMPT 1 is amessage to the subscriber which tells the subscriber: “State the name ofthe person you wish to dial”.

In step 208, the user speaks the name of the person they wish to calland the audio signal is supplied to the VD IP 18 for analysis.

Assuming the VD IP 18 recognizes the name spoken by the caller, in step209, the IP returns the called party's ID, e.g., telephone number, tothe SSP 4 which then terminates the connection to the IP 18. Optionally,in step 209, prior to returning the telephone number, the IP may playthe subscriber a confirmation message indicating the name of the partybeing called. In one embodiment, the text of the recognized name ifprocessed by TTS device 105 to generate an audio version of therecognized name which is played to the subscriber. Operation proceedsfrom step 209 to step 210, wherein the SSP 4 returns the called party ID(telephone number) to the ISCP.

In response to receiving the called party's telephone number from theSSP, in step 211, the ISCP 16 sends a Forward Call Response message withthe called party's telephone number. The ISCP also sends an O-Busy, anO-NoAnswer NEL to the SSP which will cause the SSP to signal the ISCP ifa busy or no answer condition is detected on the called party's line.

In step 212, the SSP 4 routes the call to the called party's number inan attempt to connect the caller 22 with the first called party 26. Instep 213, the SSP 4 detects whether the called party answered or ifthere is a busy, or no answer condition on the called party's line. Ifthe call is answered, the call is allowed to terminate in a normalmanner and the ISCP has no further involvement in the call.

However, in accordance with one feature of the present invention, if noanswer is detected or a busy signal is detected on the called party'sline, call processing continues in step 214 with the SSP sending a NELmessage back to the ISCP 16 along with calling party and/or callidentification information.

In response to the message from the SSP 4, in step 215, the ISCPprepares a second STOR message to be sent to the SSP 4. This time theSTOR message includes a Play APP 3 instruction if a busy signal wasdetected and a Play APP 4 instruction if a no answer condition wasdetected. The Play App 3 instruction causes the VD IP 18 to play amessage indicating that the called number is busy and asking the callerif they want to place another call. It also causes the VD IP 18 tomonitor for a response from the caller. The Play APP 4 message causesthe VD IP 18 to play a message indicating that there is no answer on thecalled line and asking the caller if they want to place another call.The Play App 3 and 4 instructions also cause the VD IP 18 to monitor fora response from the caller to the message.

In step 216, the STOR message and corresponding Play App message is sentby the ISCP 16 to the SSP 4. In step 217 an ISUP message which includesthe received PLAY APP information is sent to the VD IP 18 and the calleris once again connected to the VD IP 18.

In step 218, the VD IP 18 plays the message corresponding to the PLAYAPP message received from the SSP 4. In an exemplary embodiment, in thecase of PLAY APP2, the caller is played the message: “The number you areattempting to call is busy, would you like to place another call?”. Inthe same exemplary embodiment, in the case of PLAY APP 4, the caller isplayed the message “There is no answer at the number you are trying tocall, would you like to place another call?”.

In step 219, the caller responds to the message prompt with a “Yes”,“No” or by hanging up.

In step 220, the VD IP 18 detects the caller's response. If the callerhangs up or replies with a “No”, the call is terminated in step 221 inthe normal manner, i.e., the VD IP 18 and ISCP 16 closes records and/ortransactions associated with the call and releases resources dedicatedto servicing the call. However, if in step 220, the VD IP 18 detects a“Yes” from the caller operation proceeds once again to step 207 therebyproviding the caller the opportunity to call a subsequent, e.g., 2^(nd),party.

The dotted line associated with step 212 is used to indicate thecontacting of a subsequent called party as a result of the additionalopportunity or opportunities provided to a caller to place a voicedialing call.

The process of providing the caller the opportunity to call additionalparties through the use of voice dialing may be repeated multiple times,e.g., until a called party answers, the caller hangs up terminating thecall, or the caller indicates in step 219 that he/she does not wish tomake an additional voice dialing call.

In addition to providing the caller the opportunity to call a subsequentparty when the first called party is busy or does not answer, thepresent invention can be used to provide a caller the opportunity toplace one or more calls via voice dialing after an initial voice dialingcall is successfully completed to a called party.

In one such embodiment, when the voice dialing subscriber completes avoice dialing call to a called party, the termination of the call, e.g.,by the called party hanging up is detected by the SSP 4. The detectionof the called party hanging up is achieved through the use of an AINmid-call trigger set on the subscriber's, e.g., calling party's, line.Activation of the mid-call trigger causes the SSP 4 to launch anothermessage to the ISCP 16. In response to the message, the ISCP 16 sends aSTOR instruction and PLAY APP 5 message to the SSP 4. After beingconnected once again to the VD IP 18, the caller is played a messageinquiring whether or not the caller wants to place another call. Themessage may be, e.g. “Would you like to place another Voice Dialingcall?”. Operation then proceeds, in such an embodiment, as previouslydiscussed in regard to step 220, with the caller's response beingdetected and the caller being reconnected to the voice dialing IP 18 ifthe response was in the affirmative.

In accordance with the present invention, when a calling party hangs up,the SSP 4 sends a message to the ISCP 16 causing the ISCP 16 to closethe subscriber's CPR and completing the call transaction.

The method of providing a voice dialing service subscriber theopportunity to call additional parties without having to hang up andinitiate an entirely new call, has been described in the context of acaller placing a call from a landline phone designated as correspondingto a Centrex customer who subscribes to a voice dialing service.However, it should be noted that the voice dialing service can also beprovided to callers using mobile phones and land line phones other thantheir assigned Centrex phone.

In accordance with one Centrex embodiment, Centrex customers are allowedto designate a telephone number, other than their landline Centrextelephone number, as a Centrex mobile and/or Centrex easy accesstelephone number. The mobile and/or easy access telephone numbersassociated with a Centrex subscriber are stored as part of thesubscriber's CPR in the ISCP 16. In accordance with the presentinvention, a subscriber can access his/her Centrex service, e.g., voicedialing service, by dialing the telephone number associated with thesubscriber's landline phone.

The call to the subscriber's landline phone activates a terminatingattempt trigger (TAT) set on the subscriber's line. In response toactivation of the TAT trigger, the SCP 4 generates a request for callprocessing instructions which is sent to the ISCP 16 associated with thesubscriber. The request includes subscriber identification information,e.g., the subscriber's landline phone number, and calling partyidentification information, e.g., the calling parties telephone numberobtained using e.g., using automatic number identification (ANI)information. The ISCP 16 accesses the CPR corresponding to theidentified subscriber and determines if the call is from thesubscriber's mobile or easy access number, if it is, voice dialingservice is provided in the same manner as in the case where thesubscriber call's from his/her Centrex landline phone.

Provision of a voice dialing service to a subscriber calling from amobile phone is illustrated in FIG. 19. As illustrated, the processbegins in step 201′ with the subscriber calling his Centrex landlinephone number from mobile phone 37. The call is routed to the SSP 4 whichis coupled to the subscriber's Centrex landline phone.

In step 202′ a terminating attempt trigger set on the subscriber's lineis activated as a result of the call at the SSP 4 and call processing ispaused. In step 203′ a query is launched to the ISCP 16 for callprocessing instructions. As discussed above, the query includes callingparty identification information, e.g., the telephone number of mobilephone 37, and user (called) party, e.g., Centrex subscriber,identification information e.g., in the form of the called telephonenumber. In step 204′, using the user ID information, e.g., calledtelephone number, the ISCP 16 opens the CPR corresponding to the calledsubscriber and generates a STOR message to the IP address assigned inthe CPR to provide voice dialing service. A PLAY APP1 message, e.g.,instruction to provide a voice dialing prompt, is sent, in step 205′,along with the STOR message, to the SSP 4. To insure that the VD IP 18to which the STOR message instructs the SSP to connect the call will beable to retrieve the subscriber's voice dialing record, the callingparty ID included with the STOR message is the subscriber's Centrexlandline telephone number. In this manner the VD ID 18 is provided withthe same calling party ID whether the originating call is from thesubscriber's Centrex landline phone, mobile phone or another, e.g.,landline, phone designated as an easy access number.

Once connected to the voice dialing IP 18, the voice dialing operationproceeds as previously described in regard to FIG. 18, starting withstep 206.

Provision of voice dialing services to subscribers who are located atremote locations, without access to their remote phone or a phone whichhas been designated as an easy access phone, are also contemplated. Inone such embodiment, access to an individual subscriber is provided byhaving the subscriber call a telephone number designated for the purposeof providing voice dialing service to remote Centrex subscribers.Different telephone numbers may be designated for providing voicedialing services to different companies. Thus, it is contemplated thatthe telephone number called for voice dialing may be, e.g., an 800telephone number designated for use in providing employee's of anindividual company access to the voice dialing service which is providedas part of the Centrex service to which the company subscribes. The 800number is provisioned, at the SSP which is responsible for servicingcalls directed to the 800 number, with a terminating attempt trigger. Inaddition, a call processing record is stored in the ISCP 16 for the 800number. The call processing record includes information identifying aCENTREX DTMF IP 21 to be used for providing security in regard to callsfrom remote Centrex Subscribers.

The steps and signal flows associated with a remote voice dialingservice subscriber obtaining voice dialing service are shown in FIG. 20.

As illustrated, the process of a remote VD subscriber placing a voicedialing call begins in step 1701 wherein the remote VD subscriber callsthe telephone number, e.g., 800 number, assigned for use in providingvoice dialing service to remote subscribers. The call from the remotelocation, e.g., customer premises 28, is routed by the telephone networkto SSP 2 which is the terminating SSP for the called number. In step1702, the terminating attempt trigger set on the called number as SSP 2is hit and call processing is paused while a query requesting callprocessing instruction is prepared.

In step 1703 a query for call processing instructions is launched to theISCP 16. The query includes a user ID indicating the called number whichis designated for servicing remote voice dialing subscribers.

Using service key analysis on the USER ID included in the query, theISCP 16, in step 1704, opens the CPR associated with the called voicedialing service number. The ISCP then generates a STOR message with aPLAY APP 2 instruction and an IP address corresponding to the address ofCentrex DTMF IP 21. A Play APP 2 message prompts a caller for a CentrexID and a PIN.

In step 1705, the generated STOR message with the address of DTMF IP 21and the PLAY APP 2 instruction is sent to the SSP 2. In step 1706, theSSP 2 in response to the STOR instruction, connects the caller to theCENTREX DTMF IP 21. The connection may be local or remote to the SSP 2.

In step 1707, the IP 21 plays prompts to the caller by playing a PLAYAPP 2 messages such as: “Please enter your Centrex telephone numberfollowed by your PIN.”

In response to the prompt, in step 1708, the caller supplies thecaller's Centrex telephone number, e.g., the telephone number of phone38, assuming the remote caller is the first subscriber and a PIN. Theinput may be in the form of spoken numbers or DTMF input. In step 1709the IP 21 converts the spoken and/or DTMF signals into the correspondingnumeric values and supplies the information, e.g., Centrex phone numberand PIN to the SSP 2. The SSP 2, in step 1710 then forwards the ID andPIN information to the ISCP 16.

In step 1711, the ISCP 16, accesses the CPR corresponding to the Centrexphone number provided, compares the PIN in the CPR to received PIN. Ifthe stored and received PINS match, the ISCP identifies the address ofthe VD IP 18 assigned to service the customer and formulates a STORmessage with a PLAY APP 1 instruction and the address of the VD IP 18.However, if the ISCP 18 determines the PIN's do not match after threeattempts, the ISCP 18 instructs the SSP 2 to terminate the call, e.g.,after using the IP 10 to provide the caller with an error message. Thetransfer of the STOR message or instruction to terminate the callconnection generated in step 1711 is transmitted to the SSP 2 in step1712.

Assuming the caller was validated by the ISCP 16, in step 1713 thecaller is coupled, by the SSP 2 to the voice dialing IP 18 indicated bythe address included in the STOR message. The User ID provided to the VDIP 18, when establishing the connection with the caller is the CentrexID, e.g., subscriber's Centrex telephone number. The VD IP 18 may belocal or remote to the SSP 2 i.e., the SSP 2 may use a voice dialingplatform coupled to a different SSP, e.g., SSP4. Since the User IDprovided to the VD IP 18 is the same as the ID which would have beenprovided if the subscriber was calling from his/her Centrex phone, theVD IP 18 can access the subscriber's voice dialing information using thesame identifier regardless of how the subscriber is connected to the VDIP 18. Thus, the VD IP 18 can service the voice dialing call from theremote user in generally the same manner as a call from the subscriber'sCentrex telephone line. Accordingly, call processing in the FIG. 20embodiment proceeds from step 1713 to step 207.

In step 207, the VD IP 18 recognizes, e.g., based on the USER ID, thesubscriber just connected to the VD IP 18 and retrieves the subscriber'svoice dialing information, e.g., personal dialing directory information.The retrieved information includes speaker independent speechrecognition models of names and telephone numbers corresponding to themodeled names. Once the IP 18 is ready to perform a voice dialingoperation, as part of step 207, the VD IP 18 retrieves PROMPT 1 from itsmemory and plays PROMPT 1 to the subscriber. Prompt 1 is a message whichtells the subscriber: “State the name of the person you wish to dial”.

From step 207, the call is ultimately completed as discussed previouslyin regard to the FIG. 18 example, e.g., by following steps 209 through221 as may be dictated by received input and/or called partyavailability.

In the above described manner, a Centrex subscriber can initiate a voicedialing operation from a specific phone line, e.g., a phone lineprovisioned with Centrex service, a mobile phone, a phone correspondingto an “easy access number”, or a remote phone with which Centrex orvoice dialing service information has not been associated in recordsmaintained by the ISCP 16.

It should be noted that the FIG. 20 call flow assumes that separate IP'sare used for security and voice dialing functions. These functionscould, if desired, be implemented using the same IP.

Speech recognition models for names have been described as part of asubscriber's voice dialing record. The generation and storage of thesespeech recognition models, prior to having to perform a speechrecognition operation, reduces the amount of processing which must beperformed at speech recognition time. However, since the textinformation from which name models are generated are stored in thecustomer record, it is possible to avoid the storage of these modelsand, instead, generate them when needed from the text informationincluded in a subscriber's voice dialing record.

Numerous variations on the above described methods and apparatus arepossible without departing from the scope of the invention.

What is claim is:
 1. A voice dialing method, comprising: setting a firsttrigger on a telephone line at a telephone switch; in response toactivation of said first trigger, contacting a service control point forcall processing instructions; operating the service control point toinstruct said switch to connect a first party on said telephone line tovoice dialing circuitry; and operating the switch to disconnect thefirst party from the voice dialing circuitry after receiving saidtelephone number information from said voice dialing circuitry withoutdisconnecting said first party from the switch.
 2. The method of claim1, wherein said first party is a party attempting to initiate a voicedialing call; and wherein said first trigger is a termination attempttrigger.
 3. The method of claim 1, wherein said first trigger is avertical access trigger.
 4. The voice dialing method of claim 1, whereinoperating the service control point to instruct said switch includes:transmitting an instruction from the service control point to saidswitch; and wherein the method further comprises: operating said switch,in response to said instruction, to connect the calling party to saidvoice dialing circuitry.
 5. The method of claim 2, wherein said servicecontrol point is an integrated service control point; and whereinoperating the service control point includes: accessing one of aplurality of call processing records stored in the service controlpoint, each call processing record including call processing informationcorresponding to one of a plurality of voice dialing servicesubscriber's.
 6. The method of claim 4, wherein said instruction is partof a send to outside resource message.
 7. The method of claim 6, whereinsaid message includes an address identifying said voice dialingcircuitry.
 8. The voice dialing method of claim 4, further comprisingthe step of: operating the switch to route the call in response totelephone number information received from said voice dialing circuitry.9. The method of claim 1, further comprising: activating a secondtrigger set at the switch on said telephone line in response to theoccurrence of an event on said line; reconnecting said first party tosaid voice dialing circuitry; and operating the voice dialing circuitryto perform a speech recognition operation on speech received from saidfirst party to identify the name of an additional party to be called.10. The method of claim 9, wherein the voice dialing circuitry is anintelligent peripheral device coupled to the said telephone switch. 11.The method of claim 1, further comprising: operating the switch toestablish a voice connection between the first party and the voicedialing circuitry; retrieving voice dialing information corresponding tothe first party from a plurality of voice dialing records; and using theretrieved voice dialing information to identify a telephone number to bedialed.
 12. The method of claim 11, wherein the step of operating theservice control point includes the step of: supplying the switch with asubscriber identifier to be used when performing a voice dialingoperation.
 13. The method of claim 12, wherein the step of operating theservice control point includes the step of: performing a look-upoperation using a set of stored customer identifier information andusing a telephone number corresponding to a telephone from which thefirst party is calling, to identify a user identifier corresponding tosaid telephone number.
 14. The method of claim 13, wherein said step ofcontacting a service control point includes the step of: using automaticnumber identification information to obtain the telephone numbercorresponding to the telephone from which the first party is calling;and sending the telephone number of the first party to the servicecontrol point.
 15. A voice dialing method, comprising: setting a firsttrigger on a telephone line at a telephone switch, said first triggerbeing a mid call trigger; in response to activation of said firsttrigger, contacting a service control point for call processinginstructions; and operating the service control point to instruct saidswitch to connect a first party on said telephone line to voice dialingcircuitry.
 16. A voice dialing method, comprising: setting a firsttrigger on a telephone line at a telephone switch said first triggerbeing a feature code trigger; in response to activation of said firsttrigger, contacting a service control point for call processinginstructions; and operating the service control point to instruct saidswitch to connect a first party on said telephone line to voice dialingcircuitry.
 17. A voice dialing method, comprising: setting a firsttrigger on a telephone line at a telephone switch; in response toactivation of said first trigger, contacting a service control point forcall processing instructions; operating the service control point toinstruct said switch to connect a first party on said telephone line tovoice dialing circuitry; wherein the step of operating the servicecontrol point includes the step of: supplying the switch with asubscriber identifier to be used when performing a voice dialingoperation, the subscriber identifier being the telephone number of asubscriber's land-line phone and wherein said telephone numbercorresponding to the telephone from which the first party is calling isa subscriber's mobile telephone number, said step of supplying theswitch with a subscriber identifier including the step of correlatingthe subscriber's mobile telephone number to the subscriber's land-linetelephone number.
 18. The method of claim 17, wherein the subscribersland-line telephone number is a telephone number for which Centrexservice is provided.
 19. A voice dialing method, comprising: sending amessage to an integrated service control point, the message including afirst telephone number, the first telephone number being a mobiletelephone number; operating the service control point to use the firsttelephone number to access a call processing record; identifying, usinginformation included in the accessed call processing record a subscriberidentifier the subscriber identifier being a land-line telephone number;and supplying the identified subscriber identifier to voice dialingcircuitry.
 20. The voice dialing method of claim 19, further comprisingthe step of: operating the voice dialing circuitry to retrieveinformation in a data record associated with the supplied subscriberidentifier.
 21. The voice dialing method of claim 19, furthercomprising: operating a telephone switch to generate said message inresponse to a signal received from a telephone associated with the firsttelephone number.
 22. The voice dialing method of claim 20, furthercomprising the step of: operating the voice dialing circuitry to performa speech recognition operation to identify a name; and operating thevoice dialing circuitry to supply a second telephone numbercorresponding to an identified name to a telephone switch.
 23. A voicedialing system, comprising: voice dialing circuitry including a speechrecognition circuit for performing a speech recognition operation; atelephone switch coupled to the voice dialing circuitry; and a servicecontrol point coupled to the telephone switch, the service control pointincluding voice dialing information relating to a plurality of voicedialing service subscribers, the voice dialing information correspondingto at least one voice dialing service subscriber including: informationassociating at least two different telephone numbers corresponding totelephones used by said service subscriber to place calls with asubscriber identifier, said at least two different telephone numbersbeing suitable for comparison to a calling party number to obtain theassociated subscriber identifier from said voice dialing information.24. The voice dialing system of claim 23, wherein the subscriberidentifier is a first telephone number, the plurality of telephonenumbers including a land-line telephone number and a mobile telephonenumber associated with the same subscriber.
 25. The voice dialing systemof claim 23, wherein the voice dialing circuitry is an intelligentperipheral device including means for performing speech recognitionoperations using speak recognition models of names generated from textversions of said names.
 26. The voice dialing system of claim 23,wherein the service control point includes means for supporting aplurality of Centrex services through the use of call processingrecords, said voice dialing information being stored in at least some ofsaid call processing records.
 27. An integrated service control point,comprising: means for receiving signals from telephone switches; aplurality of call processing records corresponding to voice dialingservice subscribers, each call processing record including informationidentifying voice dialing circuitry to be used to perform a voicedialing operation when servicing the subscriber with which the record isassociated; and means for coupling the service control point to theInternet; and means for updating call processing records in response toinformation received over the Internet.
 28. The integrated servicecontrol point of claim 27, wherein at least one of the call processingrecords includes a plurality of telephone numbers associated with avoice dialing service subscriber identifier.
 29. The integrated servicecontrol point of claim 27, wherein said plurality of telephone numbersincludes at least two land-line telephone numbers and one mobiletelephone number.
 30. The integrated service control point of claim 28,wherein the call processing records include information for providing aplurality of Centrex services.